Oversampling processing, which converts the original sampling frequency to its multiple, is conventionally applied to a digital audio signal before the signal is input to a digital/analog converter. With this processing, in a digital audio signal output from the digital/analog converter, the phase characteristic of an analog anti-alias filter is maintained at a constant level in a higher-frequency zone of audible frequencies, and the effect of image noise in a digital system caused by sampling is eliminated.
In such oversampling processing, a digital filter of a linear (straight line) interpolation method is usually used. If the sampling rate is changed, or data is missing, such a digital filter obtains the average of a plurality of existing data to generate linear interpolation data.
A digital audio signal obtained after oversampling processing has a several-times-larger amount of data in the time domain due to linear interpolation, but its frequency band is not largely changed from that obtained before the conversion and its sound quality is not improved. In addition, since interpolation data is not necessarily generated according to the waveform of the analog audio signal obtained before the A/D conversion, waveform reproducibility is little improved.
When a digital audio signal having a different sampling frequency is dubbed, a sampling-rate converter is used to convert the frequency. Even in such a case, only linear data interpolation is performed by a linear digital filter, and it is difficult to improve sound quality and waveform reproducibility. In addition, the situation is the same when a data sample of a digital audio signal is missing.